[client] audio: fix underrun clock recovery

Recover stable local playback in the device clock domain and restart
rate control from the measured feed-forward clock ratio.

Ignore catch-up packet sizing until the normal source cadence returns.
This commit is contained in:
Geoffrey McRae
2026-08-22 05:50:25 +10:00
parent 5b106015de
commit d8e7966087
3 changed files with 160 additions and 56 deletions

View File

@@ -654,6 +654,54 @@ static void testPlaybackJitter(void)
CHECK(source.arrivalJitterSec < 0.011);
}
static void testPlaybackRecovery(void)
{
reset();
startAudio();
const LG_AudioFormat format = makeFormat(LG_AUDIO_FMT_S16_LE);
CHECK(playbackStart(&format, NULL, false, false));
CHECK(audio.playback.rateControl == PLAYBACK_RATE_BACKEND);
playbackSetState(STREAM_STATE_RUN);
PlaybackSourceData * source = &audio.playback.sourceData;
const int64_t arrival = nanotime();
playbackClockReset(&source->deviceClock,
arrival, 0.0, 1.0 / format.sampleRate);
source->deviceClockStable = true;
source->outputPosition = 0;
source->lastRatio = 0.999;
source->lastClockRatio = 1.0002;
source->sourcePacketDurationSec = 0.010;
playbackPublishDeviceTiming(16, arrival, 0, 0.0, 0);
CHECK(ringbuffer_consume(audio.playback.buffer, NULL, 16) == 16);
CHECK(ringbuffer_getCount(audio.playback.buffer) == -16);
uint8_t frames[1200 * 2 * 2] = { 0 };
CHECK(playbackData(frames, 1200, NULL, arrival) ==
PLAYBACK_DATA_PROCESSED);
CHECK(source->sourcePacketDurationSec < 0.010);
CHECK(source->sourcePacketDurationSec > 0.009);
CHECK(source->sourcePacketRecovering);
CHECK(source->outputPosition == 1770);
CHECK(ringbuffer_getCount(audio.playback.buffer) == 1754);
CHECK(fabs(atomic_load(&audio.playback.backendResampleRatio) -
source->lastClockRatio) < 0.000001);
CHECK(playbackData(frames, 1200, NULL, arrival + 1000) ==
PLAYBACK_DATA_PROCESSED);
CHECK(source->sourcePacketDurationSec < 0.010);
CHECK(source->sourcePacketRecovering);
CHECK(playbackData(frames, 480, NULL,
arrival + INT64_C(10000000)) == PLAYBACK_DATA_PROCESSED);
CHECK(!source->sourcePacketRecovering);
CHECK(fabs(source->sourcePacketDurationSec - 0.010) < 0.000001);
stopAudio();
}
static void testConsent(void)
{
reset();
@@ -781,6 +829,7 @@ static const struct Test tests[] =
{ "provider" , testProvider },
{ "playback-retry", testPlaybackRetry },
{ "jitter" , testPlaybackJitter },
{ "recovery" , testPlaybackRecovery },
{ "consent" , testConsent },
{ "quiesce" , testQuiesce },
};